[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 23

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

Re: [asterisk-users] ZAP device reference in Zaptel 1.4 - SIMILAR

I used the packages that were mentioned and have a link on the main website and www.asterisk.org/download I thought these were ok for production/use... I just compiled 1.4.1 (./configure and make, no make install) and copied the chan_zap.so module into /usr/lib/asterisk/modules, restarted asteri

[asterisk-users] HPEC audio clipping

I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing:

Re: [asterisk-users] Open Source VoIP client (on a webpage)

Jason Wolfe wrote: I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source cl

Re: [asterisk-users] IAX Trunk Failover

You have call-${DIALSTATUS} and s-CONGESTION. It might not be CONGESTION. Do a Noop(${DIALSTATUS}) you should get something. Justin On 4/5/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote: tried x+102 ? On 4/5/07, Brent <[EMAIL PROTECTED]> wrote: > I'm trying to get an IAX trunk to failover

Re: [asterisk-users] Asterisknow or Trixbox?

I know you will receive many replies to your request. We have selected Thirdlane PBX Manager (www.thirdlane.com) to manage our Asterisk installations. The key for us was a management system that was modest in cost and allowed us to easily provide customizations when necessary, yet allow the s

Re: [asterisk-users] Call dies when I press *

Man, I really thought you had nailed it. So I defined the variable that holds those paramters to the empty string and tried again. From my console log: Dial("SIP/line_3-b7701f30", "IAX2/Uxxx/18003310500|90|") in new stack I still have the same symptoms. However, if I dial 5058457900, I'm

Re: [asterisk-users] PRI DCHAN Errors

On 4/5/07, Rob Schall <[EMAIL PROTECTED]> wrote: Yes, the connection seems solid and the cable is alright. That doesn't mean the cable is the proper specification. Most people use category 5 unshielded twisted pair which technically is not the correct cable. While this is probably not the ca

[asterisk-users] Open Source VoIP client (on a webpage)

I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggesti

[asterisk-users] BeroNet HFC-4S card is now detected as only 2 ports

Hello list, After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently detected as 2 ports instead of 4. I still load the driver as "modprobe qozap ports=12" as I've always done. But now it only sees 2 ports. Output of "lspci -vvv" -- cut 02:01.0 ISDN controller: Cologne Ch

RE: [asterisk-users] Analog phones, dial out

Paste here the rules of your extensions.conf for outgoing calls. Sds, Gustavo From: "Joe Acquisto" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Subject: [asterisk-users] Analog phones, dial out Date: Thu, 05 Apr 2007 16:28:44 -0400 I have a sys

[asterisk-users] Re: ZAP device reference in Zaptel 1.4

Hi, On Wed, Apr 04, 2007 at 04:48:08PM +0300, Tzafrir Cohen wrote: > On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: > > On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: > > > monk*CLI> zap show channels > > > No such command 'zap show' (type 'help' for help) > > >

Re: [asterisk-users] Asterisknow or Trixbox?

I also am curious to hear feedback on either asterisknow and/or trixbox. I am looking to install a few boxes in locations that people need to be able to add extensions (queues, etc), but might not be avid linux users. I could write my own gui, but why bother. A nice easy installation and extension/

Re: [asterisk-users] Asterisknow or Trixbox?

On Thu, 5 Apr 2007, WipeOut wrote: I am sure its been discussed before but I couldn't find it in my searches.. Looking to replace my Asterisk box (Ver 1.0 still I think) and really like the idea of an easy to use gui to manage it.. I see the contenders appear to be Asterisknow and Trixbox..

RE: [asterisk-users] Queue call distribution

If you set the queue strategy to ringall it should ring all the interfaces you have set up in that queue. Just make sure you have member => SIP/EXT setup. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Thursday, April 05, 2007 4:06 PM To: asterisk-users@lists

Re: [asterisk-users] Queue call distribution

Jordan Novak wrote: > I have noticed that asterisk will only try one interface per queue at a > time. Is there any way get get it to dial say three at a time and > connect the first one that it reaches. Do you have autofill=yes in queues.conf? Regards, Philipp -- amooma GmbH - Bachstr. 126

[asterisk-users] Analog phones, dial out

I have a system with a TDM400p 2FXO, 2FXS. Analog phones work fine, on incoming calls, ring, answer, talk, hangup. However, not so good dialing out. Pickup handset, get dail tone. "Cli shows Starting simple switch on Zap/1-1". Press a key and get "Hungup Zap/1-1" and get the "doot-doot-doot"

Re: [asterisk-users] PRI DCHAN Errors

Andrew Joakimsen wrote: > On 4/5/07, Rob Schall <[EMAIL PROTECTED]> wrote: >> Hey all, >> >> I had a user complaining of calls which were dropping mid-conversation. >> I looked into the time of one of the calls, and saw the following: >> >> Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels

[asterisk-users] Queue call distribution

I have noticed that asterisk will only try one interface per queue at a time. Is there any way get get it to dial say three at a time and connect the first one that it reaches. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mai

[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 22

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

[asterisk-users] IAX2 threads

I've been running an asterisk box that is routing an average of 9 simultaneous calls between a PRI and five asterisk boxes via IAX. After having it run for about five hours I had a spate of these error messages: Out of idle IAX2 threads for I/O, pausing! And then they went away. The only reference

[asterisk-users] Asterisknow or Trixbox?

I am sure its been discussed before but I couldn't find it in my searches.. Looking to replace my Asterisk box (Ver 1.0 still I think) and really like the idea of an easy to use gui to manage it.. I see the contenders appear to be Asterisknow and Trixbox.. Has anyone player with both who can

Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12

Since when is Canada part of the "rest of the world?" I thought it was a US National Park? ;-) On 4/5/07, Stephen Bosch < [EMAIL PROTECTED]> wrote: john beaman wrote: > I too was curious about this, so I copied the text into Babel Fish, and this is the result: > > I miss of the 2/04/2007 to t

Re: [asterisk-users] Polycom and Asterisk

Stephen Bosch wrote: Andrew Joakimsen wrote: Well I would wonder how Polycom even had any idea whom your vendor is. The vendor made a request for 2.1.0 on my behalf and let it slip that it was for one of my clients :) What is it about current firmware that makes them so paranoid? For

Re: [asterisk-users] PRI DCHAN Errors

On 4/5/07, Rob Schall <[EMAIL PROTECTED]> wrote: Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-c

RE: [asterisk-users] disabling authentication

From: "Mark Price" <[EMAIL PROTECTED]> Date: Wed, 4 Apr 2007 10:07:31 -0400 Is there a way to cause asterisk to accept all calls without any authentication? Mark Yes - not to set up a user/peer section in sip.conf. The context in [general] section will be used. Yuan Liu

[asterisk-users] Dialplan not reading MySQL table

Hello, I'm trying to use MySQL for Dialplans and have followed the Asterisk RealTime Extensions setup. The MySQL table is called "extensions" and I have entered two records.. ext 1000 and 2000. I also added switch => Realtime/[EMAIL PROTECTED] in extensions.conf and extensions => mysql,aster

Re: [asterisk-users] PRI DCHAN Errors

On 4/5/07, Rob Schall <[EMAIL PROTECTED]> wrote: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.

Re: [asterisk-users] Polycom and Asterisk

Maybe the firmware uses GPL'd code? ;-) Just a theory, don't sue me Polycom! On 4/5/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: Andrew Joakimsen wrote: > Well I would wonder how Polycom even had any idea whom your vendor is. The vendor made a request for 2.1.0 on my behalf and let it slip th

Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12

john beaman wrote: > I too was curious about this, so I copied the text into Babel Fish, and this > is the result: > > I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of > my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric > Buzay. > > If this guy

[asterisk-users] detecting a "beep"

at the moment, if our agents make a call and they get an answering machine they have to wait for the "beep" before leaving a message. I would like them to be able to transfer the call to an extension where an automated message can be left as soon as they know it is an A/M. However, how do I kn

Re: [asterisk-users] Best Hardphone (Subjective?)

Michael Graves wrote: > I think that the Polycom phones are really good, but their lack of > support for Asterisk is legendary. Though I'm no fan of some of Polycom's policies, this isn't strictly true anymore. They know which way the wind is blowing: http://forms.polycom.com/audio_files/techpart

Re: [asterisk-users] IAX Trunk Failover

tried x+102 ? On 4/5/07, Brent <[EMAIL PROTECTED]> wrote: I'm trying to get an IAX trunk to failover to a local trunk it the trunk is down. This is what I've been working on: [macro-forward1]; exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20) exten => s,2,Goto(call-${DIALSTATUS},1) exten

Re: [asterisk-users] Best Hardphone (Subjective?)

Bill Hackensack wrote: > On 4/2/07, *Corporate IT Solutions - Michael Dunne* > <[EMAIL PROTECTED] > wrote: > > So subjectively what would be the best Hardphone for a small/medium > business with multiple line support, BLF, etc. > > > Does _anyone_ read the arch

[asterisk-users] IAX Trunk Failover

I'm trying to get an IAX trunk to failover to a local trunk it the trunk is down. This is what I've been working on: [macro-forward1]; exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20) exten => s,2,Goto(call-${DIALSTATUS},1) exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20) exten => s-CHANUNA

Re: [asterisk-users] What is this error message? (check_auth: stale nonce received from ...)

actually i think it's ... stale nonce stale as in old.. nonce is auth related.. -- Forwarded message -- From: Mike Lynchfield <[EMAIL PROTECTED]> Date: Apr 5, 2007 12:44 PM Subject: Re: [asterisk-users] What is this error message? (check_auth: stale nonce received from ...) To

Re: [asterisk-users] What is this error message? (check_auth: stale nonce received from ...)

iirc.. check_auth: stale nonce received from ' is.. asked to auth but auth expiry still good..continue.. On 4/5/07, Mike <[EMAIL PROTECTED]> wrote: I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received f

Re: [asterisk-users] Polycom and Asterisk

Andrew Joakimsen wrote: > Well I would wonder how Polycom even had any idea whom your vendor is. The vendor made a request for 2.1.0 on my behalf and let it slip that it was for one of my clients :) What is it about current firmware that makes them so paranoid? For pete's sake! If the argument is

[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 21

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

Dovid B wrote: I have created this before. I have to dig up the dial plan. The way I created it is it would call user1. User1 had the option to take the call, pass it to the next user or send it to VM. If he passed it to the next user, User2 had the same options as user1 and it flows down the lis

Re: [asterisk-users] Configuration assistance needed.

What is locally ? Where are you located ? - Original Message - From: "Tim King" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, April 05, 2007 6:15 PM Subject: [asterisk-users] Configuration assistance needed. I have a fairly lar

Re: [asterisk-users] polycom repair

Jesse, I would just add that if he has a 600 it's worth getting the 601. The 600 was limited in memory and had some other disadvantages. James I would hang on to the phone for testing or something else. Even though there may be no screen, you can program it via the web interface and it can be us

Re: [asterisk-users] Polycom and Asterisk

> It's good > to know Polycom has anti-competitive business practices. I also > dislike that they refuse to give out anything but old firmware > versions too. They could do a lot to improve their relationships with their public :( I second that! I like the 501 and 601 phones I have, and they c

RE: [asterisk-users] Polycom 601 message waiting indicator

Thanks David and Chris, appreciate the response Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Thursday, April 05, 2007 11:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 60

Re: [asterisk-users] Polycom 601 message waiting indicator

Mike wrote: Hi, I'm used to the Polycom 501, with a big message light indicator that flashes when I have a message waiting. As far as I can see when looking at pictures, the 601 (and 650) do not have this indicator light (nor does the 550 for that matter). How does it show the user that i

[asterisk-users] No response on extensions: TDM842

Hi all I recently bought a TDM842, basically a TDM800P card with 4 FXS, 2 FXO's. Installed the same driver as for the TDM24xx cards, as the manual says. Running [EMAIL PROTECTED] 2.8, did yum update, now on kernel 2.6.9-42.0.10.ELsmp Card config: Channel 1-4: FXS, Channel 5,6 FXO >From dmesg: Af

Re: [asterisk-users] "remote" SIP, no audio, or one way audio.

Steve Totaro <[EMAIL PROTECTED]> Wrote: 4/4/2007 8:44 PM: > Joe Acquisto wrote: >> Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite >> softphones, for eval/testing. They do get registered, and can call each >> other, but mostly get no audio, sometimes one way audio. >> >

[asterisk-users] What is this error message? (check_auth: stale nonce received from ...)

I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from ' I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike ___

[asterisk-users] Configuration assistance needed.

I have a fairly large system to configure. I was hoping to find someone locally to employ for this project but remote configuration is considerable. Pleas let me know if you are interested and have the time. ___ --Bandwidth and Colocation provided by Eas

Re: [asterisk-users] Polycom 601 message waiting indicator

It has one, you just can't see it as easily in photos. It is to the right top corner of the display, top edge of the phone. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This mess

[asterisk-users] Polycom 601 message waiting indicator

Hi, I'm used to the Polycom 501, with a big message light indicator that flashes when I have a message waiting. As far as I can see when looking at pictures, the 601 (and 650) do not have this indicator light (nor does the 550 for that matter). How does it show the user that it has a message

Re: [asterisk-users] Call dies when I press *

Is it related to Dial() options: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * Perhaps it is some other system inline with yours that has this feature enabled. I certainly found this to be t

Re: [asterisk-users] SDP bug

5 apr 2007 kl. 13.04 skrev Raj Jain: Regarding project Pineapple, I'm curious why rewrite (or refactor) the SIP stack instead of using an open-source one. Did your research show that there is nothing viable out there that'll fit well w/in Asterisk? OpenPBX community is talking about using

[asterisk-users] PRI DCHAN Errors

Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] chann

Re: [asterisk-users] polycom repair

James Andrewartha, We send these into Polycom for repair on occasion. It's will cost roughly $200~$300 direct with Polycom and will take 8~10 weeks for repair. Obviously, most people opt not to do a repair and just buy a new phone instead. The only time it may make sense to repair an out of

Re: [asterisk-users] "remote" SIP, no audio, or one way audio.

"J. Oquendo" <[EMAIL PROTECTED]> Wrote: 4/5/2007 6:47 AM: > Joe Acquisto wrote: >> >> >> Thanks. And this might go where, in rc.d/rc.firewall.local ? >> >> But I don't get it. Isn't this redundant? Since I have port forwarding >> already. . .? >> >> joe a. >> >> __

Re: [asterisk-users] SDP bug

Olle, Regarding project Pineapple, I'm curious why rewrite (or refactor) the SIP stack instead of using an open-source one. Did your research show that there is nothing viable out there that'll fit well w/in Asterisk? OpenPBX community is talking about using Sofia-SIP stack, for instance. Raj

Re: [asterisk-users] Asterisk 1.2.17 and BRIstuff

Dominik Zalewski wrote: > I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is > that current version of BRIstuff is for Asterisk 1.2.14. > > BRIstuff 0.3.0-PRE-1y (* 1.2.14) > > If I'm misunderstanding how to apply patches for 1.2.17? Download Bristuff 0.3.0-PRE-1y-e: ht

[asterisk-users] Asterisk 1.2.17 and BRIstuff

Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik

Re: [asterisk-users] "remote" SIP, no audio, or one way audio.

Joe Acquisto wrote: Thanks. And this might go where, in rc.d/rc.firewall.local ? But I don't get it. Isn't this redundant? Since I have port forwarding already. . .? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-user

[asterisk-users] Asterisk 1.2.17 and BRIstuff - SOLVED

>Hi All, > >I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is >that current version of BRIstuff is for Asterisk 1.2.14. > >BRIstuff 0.3.0-PRE-1y (* 1.2.14) > >If I'm misunderstanding how to apply patches for 1.2.17? > > >Thank you in advance, > >Dominik I found it:) http

[asterisk-users] Re: Correct latency values in "sip show peers"

Eric "ManxPower" Wieling wrote: The times shown are the time to get a response to a SIP OPTIONS packet sent to the phone, not the time to get a response from an ICMP ECHO (ping) packet. What's the difference between yours and mine mail? -- Tomislav Parcina [EMAIL PROTECTED]

[asterisk-users] SNOM and Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip

Hi, I have an Asterisk 1.2.9.1 box and a bunch of snom phones. I sometimes get this error: *ERROR[31201] chan_sip.c: Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip* It seems that my SIP phone is sending subscribe command for numbers not inserted inside th

Re: [asterisk-users] Asterisk server hangs on after only few hours again.

johnny_xing wrote: hi, everyone, i have been sufferred for the asterisk hang on problem for so long and i just reinstalled the whole thing yesterday, but again this morning the server hangs on again, you could not call in through PSTN line and the ppl also could not call out throught the server,

[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 20

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

Re: [asterisk-users] SDP bug

3 apr 2007 kl. 10.04 skrev kjcsb: The call that gets dropped had a retransmission of INVITE from UAC to UAS (and therefore retransmission of 200 OK from UAS to UAC). There is nothing wrong with the re-transmission as such, but I noticed a potential bug in Asterisk in the way it responds to an I

Re: [asterisk-users] SDP bug

3 apr 2007 kl. 09.07 skrev Raj Jain: Olle, It depends on how strictly the UA adheres to the offer/answer model. The issue would be that a RE-INVITE from Asterisk will have the version number incremented by more than one, which will break the following rule. Quoting from RFC 3264 Sectio

Re: [asterisk-users] Red alarms

On 2/8/07, Wayne Jensen <[EMAIL PROTECTED]> wrote: On 2/8/07, Don Pobanz <[EMAIL PROTECTED]> wrote: > > Asterisk is getting red alarms on my T1, sometimes once or twice a > > day, but today it happened 5 times. Even once is too many. Every > > call in progress is dropped. > > Red alarm means th

[asterisk-users] How to return dialstatus of second (sub) call

Hello all I have this problem, i need a way to balance my "trunks" which are SIP peers, when a SIP peer is busy then send the call for another peer and so until i can send away the call, i think i can do it with queues. Ok this is the scenario: In extensions.conf [balance] exten => _,1,